Hi Chris
I have been reading about a very big horn-setup in Hamburg on a German forum. Fully horn-loaded, stereo horn-subs and three sets of five-way hornspeakers in a trinaural setup. It is a bit of a playground/developing place for Uli Brueggeman, the creator of Acourate, as he is deeplly involved in doing the filtering and developing new capabilities in his software to make things possible. It´s all active with I think 17 output-chanels, one for each horn. I don´t know about input channels on that system other then stereo.
What I want to say is: I would think that you can "handle 6 input channels and ~12 outputs" at 96kHz and as many taps as you want, with Acourate? Uli Brueggemann is very helpful, so maybe you could send him an E-mail and ask if his software can do that? And then tell us what you found out!
Here is a link to Acourate: https://www.audiovero.de/en/acourate.php
And here a link to his software-convolver, where I would think the 6 input channels and ~12 outputs should de routed: https://www.audiovero.de/en/acourateconvolver.php
Steffen
And you answered:
I have been reading about a very big horn-setup in Hamburg on a German forum. Fully horn-loaded, stereo horn-subs and three sets of five-way hornspeakers in a trinaural setup. It is a bit of a playground/developing place for Uli Brueggeman, the creator of Acourate, as he is deeplly involved in doing the filtering and developing new capabilities in his software to make things possible. It´s all active with I think 17 output-chanels, one for each horn. I don´t know about input channels on that system other then stereo.
What I want to say is: I would think that you can "handle 6 input channels and ~12 outputs" at 96kHz and as many taps as you want, with Acourate? Uli Brueggemann is very helpful, so maybe you could send him an E-mail and ask if his software can do that? And then tell us what you found out!
Here is a link to Acourate: https://www.audiovero.de/en/acourate.php
And here a link to his software-convolver, where I would think the 6 input channels and ~12 outputs should de routed: https://www.audiovero.de/en/acourateconvolver.php
Steffen
I have been thinking long time to switch to computer based DSP due to much bigger capabilities.
But never found friendly user system. When I said friendly user I am not talking about measurements and adjusting etc. I do not care about it, will find one or another way. I am talking about every day usage.
I would like to have 2x4 (5) way system with one remote and multiple inputs. I want to connect TV, play BT, play WIFI from NAS. And should be simple for use as any receiver.
What would be configuration with all those features?
But never found friendly user system. When I said friendly user I am not talking about measurements and adjusting etc. I do not care about it, will find one or another way. I am talking about every day usage.
I would like to have 2x4 (5) way system with one remote and multiple inputs. I want to connect TV, play BT, play WIFI from NAS. And should be simple for use as any receiver.
What would be configuration with all those features?
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In the mean time, I was able to connect to the DSP-408 via Bootcamp, with a new cable. In a side conversation with @jpbturbo, Josh explained how he hacked the settings to implement @Cask05 ‚s zeroth order filters, setting the pass filters below and above the natural limits of the drivers, implementing the shelf eq‘s as advised. I’ll have to double check my eq settings, but I think I’ve got it sorted. Unfortunately, my tinnitus is really bad today, so any sort of critical listening is out until hopefully tomorrow.
Or just shoot an email to Mitch - I’m sure he would be happy to advise whether Acourate or AudioLense have the right functionality.Hi Chris
And you answered:
I have been reading about a very big horn-setup in Hamburg on a German forum. Fully horn-loaded, stereo horn-subs and three sets of five-way hornspeakers in a trinaural setup. It is a bit of a playground/developing place for Uli Brueggeman, the creator of Acourate, as he is deeplly involved in doing the filtering and developing new capabilities in his software to make things possible. It´s all active with I think 17 output-chanels, one for each horn. I don´t know about input channels on that system other then stereo.
What I want to say is: I would think that you can "handle 6 input channels and ~12 outputs" at 96kHz and as many taps as you want, with Acourate? Uli Brueggemann is very helpful, so maybe you could send him an E-mail and ask if his software can do that? And then tell us what you found out!
Here is a link to Acourate: https://www.audiovero.de/en/acourate.php
And here a link to his software-convolver, where I would think the 6 input channels and ~12 outputs should de routed: https://www.audiovero.de/en/acourateconvolver.php
Steffen
Chris, thank you for your patience, I've attempted to adjust based on your parameters and I feel like I've made some progress, but I also feel like I've been chasing my tail for the last hour or so, trying to get the quasi flat phase alignment you've alluded to. I'm far from a dummy, but I've got an artist's brain, not an engineer's and this is making me feel more than a little dense, like I'm only groking about 25% of what you're putting out. I'm feeling a little in the weeds.
here is my SPL and Phase measurements after zeroing out the IR:
here's the group delay:
crossover points are at 400Hz and 6kHz, these are my settings for the shelf and PEQ filters, I think I may be a little ham-fisted with the application:
again, thank you for your patience and please accept my apologies for being thick-headed.
Lance
That phase curve below 600 Hz shows that you've likely added a bit too much delay to the higher frequency channels, since the phase line is going negative below that frequency.
In order to see this for sure, go to the group delay plot, then zoom way in on the vertical axis until you can see 1 millisecond minor divisions (1 ms) on the vertical axis scale, then go over to the controls menu on the upper right above the plot, then select "generate minimum phase" and then "generate" or "generate and close". Your group delay and excess group delay plot (making sure to check the plot box below the graph in order to see this curve) will now be able to show you directly your time/phase alignment of the different ways, sort of like this:
The above excess group delay plot (white line) shows a 3.56 ms delay from high frequency to low frequency (red markup lines). That means I would need to crank in an additional 3-3.5 ms of delay on the higher frequencies than I had when I took this measurement. In your case, the white line probably goes below the 0 (zero) ms line by about 1-2 ms, in which case you'd remove that amount of delay from the higher frequency channel(s), or if you don't have any or enough HF delay to subtract from, then add delay equal to the bottom of the excess group delay horizontal tangent line value to the LF channel.
Now this is all assuming that you're crossing over at about 600 Hz. If not, then something else is occurring.
Chris
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I'm crossing at 400Hz, so something else may be going on there. I won't be able to measure again until Wednesday, when my spouse gets to the office and I have a few hours by myself after work. I'm somewhat concerned about the spike at 400Hz; the excess delay below that is likely caused by the passive radiator mounted in the back of the upper enclosure. I'm not particularly disturbed by that, it's a small price to pay for a LaScala clone that extends to the low 20's, better than a K-horn. I would, however, like to better manage what's going on between 150-600 Hz, if possible, as well as tame the excess phase.
Thanks again,
Lance
Thanks again,
Lance
The transfer function disturbances (amplitude and phase) usually are measured above or below the actual crossover frequency, so that's not a surprise.
The fact that it's a negative excess group delay spike says that it's likely an internal reflection or a reflection by something nearby--at ~17 inches from the bass bin centerline (i.e., half-wave bounce peak, as opposed to a quarter wave bounce notch).
Excess group delay spikes are the primary way that you can separate reflections from minimum phase driver/enclosure resonances. You can't EQ out peaks or notches if the excess group delay curve shows non-flat behavior at the frequency of interest--a little tidbit I picked up from D'Appolito's little Testing Loudspeakers paperback, 1999, Audio Amateur Press.
Chris
I spent some time yesterday with my setup until the kids got tired of hearing the bwoooooeeeeep from REW.
I definitely still have a phase shift between the drivers, but much less than I had on my old measurements.
and the GD screen.
I was crossing over at 400 and 2000.
Already the kids have all mentioned that it sounds clearer/more focused than before.
Anyway,
Same DSP as @gigantic , DJK ported LS with PRV D2200PH driver on WG45-50 midhorn and Selenium D220TI on Faital Pro STH100 for the tweeter.
Cheers,
Josh
I definitely still have a phase shift between the drivers, but much less than I had on my old measurements.
and the GD screen.
I was crossing over at 400 and 2000.
Already the kids have all mentioned that it sounds clearer/more focused than before.
Anyway,
Same DSP as @gigantic , DJK ported LS with PRV D2200PH driver on WG45-50 midhorn and Selenium D220TI on Faital Pro STH100 for the tweeter.
Cheers,
Josh
The definitive test: young or female ears can usually beat us "old guys" in the hearing department (i.e., any male over the age of 30). I always look at my wife and ask her if she hears anything different--then let her tell me what's different. She nails it every time.
Chris
My hearing is impaired from growing up on a farm in Iowa, 2 years in an 8” army artillery unit, playing in bands, 30 years in the construction trades and the better part of the previous decade working in professional motorcycle racing and motorcycle track days. I know how to listen, but my ears have a mass roll-off around 9750 Hz. My spouse, otoh, has nearly perfect hearing and I’ve been teaching her to listen and serve as the final arbiter of any mods I do.
I used to work with a bunch of former artillery officers (most were O-5 or O-6, and one 3-star). Basically all of them had hearing aids in one or both ears (or needed them). MLRS and its other munitions don't have the same "bang" as the 155s and 203s. Fortunately, I wasn't exposed to any of that.
I can still hear the highs up to ~16-17 kHz--based on my work on my setup and in demastering music tracks, but it has to be louder than the headphone hearing tests used to plot hearing acuity, i.e., 80 dB max.
Chris
I was a canon fire direction specialist, I calculated firing data for 8", nuclear capable artillery. we were ultimately replaced by MLRS after the first Gulf war. I had an opportunity to go to the DoD Language institute and do MI, but that would have been a 4 year commitment and I had it in my head that I'd do two years and go back to college...
Mark this may be a dumb question but how do you control volume using Dante? I'm considering going the Dante route at some point but can't quite wrap my head around how it would work in a home theater environment. On top of that, this has to be useable by the whole family and eventually have Dolby Atmos support... preferably with the remote that came with our Apple TV 4k. My use case needs 16 channels of output (active MEH's in front + 3 subs + overheads + surrounds)... will I need a 16 channel pre-amp with a master volume control?
Fizzylime,
I'm not Mark but i use Dante too ( in a way smaller setup than Mark's though).
Dante is only a communication protocol between gear, in other words some digital 'pipes' on which audio travel from source to any destination on the matrix.
For volume control you have to rely on software ( or hardware) digital volume control capability.
As i come from pro field i'm used to DAW to do such things as they open possibility to gang ( internally) multiple volume control to one only virtual ( or physical) fader/potentiometer.
Regarding flexibility of control you won't find better than this afaik. The drawback: can be a steep learning curve and forget about family use if you've got small kids or a non geek friendly partner.
I've been struggling with that for almost 10 years now, my older son got it, my girlfriend... never! In a way it's reassuring to me as the system can output high spl and could damage their hearing.
I'm not Mark but i use Dante too ( in a way smaller setup than Mark's though).
Dante is only a communication protocol between gear, in other words some digital 'pipes' on which audio travel from source to any destination on the matrix.
For volume control you have to rely on software ( or hardware) digital volume control capability.
As i come from pro field i'm used to DAW to do such things as they open possibility to gang ( internally) multiple volume control to one only virtual ( or physical) fader/potentiometer.
Regarding flexibility of control you won't find better than this afaik. The drawback: can be a steep learning curve and forget about family use if you've got small kids or a non geek friendly partner.
I've been struggling with that for almost 10 years now, my older son got it, my girlfriend... never! In a way it's reassuring to me as the system can output high spl and could damage their hearing.
Thanks for the info Krivium! That clears things up a bit. In a way it's kind of the answer I wanted to hear as now I get to think about how to DIY a 16 channel pre-amp with a master volume that can be controlled via IR. A subject for another thread but I wonder if the Iron Pre modules available (?) from our store could be made to work in such a way?
Hi fizzylime, not a dumb question at all....lots to learn, wade into, putting together a full system.
Dante is just a data transfer method, like USB, or AES67, etc. These do nothing to control volume, which is left up to the devices connected via any of these methods.
If you are considering the Q-Sys Cores, think of them as a configurable processor, or a configurable preamp if you like.
Within a Core's physical hardware I/O capability, you tell the Core how many inputs and outputs you need, you build the input switching, routing, speaker processing if any, etc.
I do not know how I would go from Apple TV 4K HDMI into Qsys. If the Core were an110f, which has USB, maybe an HDMI to USB converter..but dunno...I don't try to integrate video with main audio system. Qys does have devices which accept HDMI directly, but not seen used very often, so $$ new.
Qsys does integrate with Dolby Atmos. https://www.qsc.com/resource-files/applicationguides/cinema/q_cin_appguide_q-sys_atmos.pdf
The Apple TV would be the volume control, once implemented.
As far as whole family, I image switching between sources would be the major first need. Qsys allows one to build whatever controls they want into a custom remote GUI, and put that GUI in a PC or IOS device as a remote. I use a touchscreen laptop.
A Core 110f can be configured for 16 analog output channels, and 8 analog input channels. Larger Cores, like 250i, 500i, and 510i, use input and output cards that plug in to provide channel count.
A software Dante license is available for the 110f. The larger Cores need a Dante plug in card. If buying a larger Core used off ebay, it's important that it have the I/O cards in it that work for you, cause they can be a $ pain to scrounge up individually.
A $50 license fee per machine, puts Dante Virtual sound card on a PC or MAC. But, there still must be a Dante host device, like a Qsys core.
Dante is just a data transfer method, like USB, or AES67, etc. These do nothing to control volume, which is left up to the devices connected via any of these methods.
If you are considering the Q-Sys Cores, think of them as a configurable processor, or a configurable preamp if you like.
Within a Core's physical hardware I/O capability, you tell the Core how many inputs and outputs you need, you build the input switching, routing, speaker processing if any, etc.
I do not know how I would go from Apple TV 4K HDMI into Qsys. If the Core were an110f, which has USB, maybe an HDMI to USB converter..but dunno...I don't try to integrate video with main audio system. Qys does have devices which accept HDMI directly, but not seen used very often, so $$ new.
Qsys does integrate with Dolby Atmos. https://www.qsc.com/resource-files/applicationguides/cinema/q_cin_appguide_q-sys_atmos.pdf
The Apple TV would be the volume control, once implemented.
As far as whole family, I image switching between sources would be the major first need. Qsys allows one to build whatever controls they want into a custom remote GUI, and put that GUI in a PC or IOS device as a remote. I use a touchscreen laptop.
A Core 110f can be configured for 16 analog output channels, and 8 analog input channels. Larger Cores, like 250i, 500i, and 510i, use input and output cards that plug in to provide channel count.
A software Dante license is available for the 110f. The larger Cores need a Dante plug in card. If buying a larger Core used off ebay, it's important that it have the I/O cards in it that work for you, cause they can be a $ pain to scrounge up individually.
A $50 license fee per machine, puts Dante Virtual sound card on a PC or MAC. But, there still must be a Dante host device, like a Qsys core.